Repo created

TMessagesProj/jni/voip/webrtc/net/dcsctp/socket/capabilities.h

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+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef NET_DCSCTP_SOCKET_CAPABILITIES_H_
+#define NET_DCSCTP_SOCKET_CAPABILITIES_H_
+
+#include <cstdint>
+namespace dcsctp {
+// Indicates what the association supports, meaning that both parties
+// support it and that feature can be used.
+struct Capabilities {
+  // RFC3758 Partial Reliability Extension
+  bool partial_reliability = false;
+  // RFC8260 Stream Schedulers and User Message Interleaving
+  bool message_interleaving = false;
+  // RFC6525 Stream Reconfiguration
+  bool reconfig = false;
+  // https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-zero-checksum/
+  bool zero_checksum = false;
+  // Negotiated maximum incoming and outgoing stream count.
+  uint16_t negotiated_maximum_incoming_streams = 0;
+  uint16_t negotiated_maximum_outgoing_streams = 0;
+};
+}  // namespace dcsctp
+
+#endif  // NET_DCSCTP_SOCKET_CAPABILITIES_H_